RF base stations typically have been used in the past to provide a link between a single land-based dispatcher and one or more mobile (or portable) RF transceivers. However, it has become desirable to permit multiple geographically separated dispatchers to simultaneously communicate with mobile/portable transceivers over the same RF channel.
Taking a fire department communications system as an example (see FIG. 1A), a first dispatcher or other communications participant (e.g., remote A as shown in FIG. 1A) may be located in a fire station; and a second dispatcher or other communications participant (remote B in FIG. 1A) may be located at a local government "911" telephone dispatch facility. The first dispatcher/communicator (remote A) and the second dispatcher/communicator (remote B) can each monitor communications carried by the RF base station, and either dispatcher/communicator can participate in the communications (e.g., to talk to mobile transceiver C) by depressing a PTT (push-to-talk) switch and speaking into a microphone. Moreover, the first and second dispatchers can also talk to one another. Thus remote dispatcher A can talk to remote dispatcher B and vice versa; and each of the remote dispatchers A and B can talk with transceiver C. This connection is known as a "parallel remote configuration."
In the parallel remote configuration shown in FIG. 1A, it is important that each communications participant hears all other communications participants at equal levels. However, for reliability, cost and other reasons (e.g., convenient conveyance of supervisory control signals not available to RF users), land lines are typically used to convey audio between the base station and the multiple dispatch/communicator locations. Thus, it has become necessary to connect multiple audio land lines (e.g., one for each dispatcher or other land-based communications facility) to a single radio base station--and to mix or sum the audio on the multiple lines for transmission over the RF channel. This multiple input arrangement has led to certain level equalization problems which have been solved in the past by using an audio compressor.
For reliable communications and intelligibility over the RF channel, it is essential that the base station average RF modulation level be kept at a relatively high percentage. So long as only one land line is connected to the base station, audio signal level adjustment is a relatively easy affair. A fixed amount of gain (active amplification) can be applied to the line signal (at the dispatcher end, the base station end, or at both ends). Longer lines typically exhibit more loss and therefore require more gain--but the amount of loss in a dedicated land line remains substantially constant over time.
However, different telephone lines may exhibit different loss characteristics. There can be a tremendous signal loss in a telephone line--and this signal loss can depend on a number of complex factors (e.g., the length of the telephone line, the amount of loading on one or both ends of the line, etc.) The line audio received by the base station may have a low level if the line exhibits a large loss. In the alternative, the line audio level may be relatively high if there is low line loss (such as where there is a short distance between the base station and a dispatcher location). Making several different lines all have the same loss is possible, but may be expensive and may require costly and specialized attenuators or active components for each line.
Anyone who has tried to talk on the telephone simultaneously with a long distance caller and someone on a local extension knows the frustration involved in trying to hear and understand a weak signal and a louder signal at the same time. Similarly, the radio dispatcher does not want to hear faint audio from, for example, a RF mobile unit and loud audio from, for example, another dispatcher. Likewise, the RF transceiver user should not have to turn his speaker volume down to listen to one talker and turn the volume up to listen to another talker.
It has long been known to use audio compression and limiting to increase the average modulation level of a RF carrier and to provide a relatively uniform audio output level that is substantially unaffected by variations in input level.
For example, broadcast stations have long used compressor/limiter amplifiers to boost average modulation level and prevent overmodulation. Typical audio sources may provide wide dB or more dynamic range (e.g., the so-called "noise floor" can be 80 or 90 dB "down" from the maximum output amplitude that can be provided without clipping). Even a standard microphone has a relatively large dynamic range. Typical RF transmitting equipment may offer much more restricted audio dynamic range.
Compression amplifiers "compress" the dynamic range(s) of the audio source(s) so that the output dynamic range "fits" into the available dynamic range provided by the transmitting and audio processing equipment. Moreover, some transmitting stations use an even greater degree of audio compression in order to make their signals "louder" on the average, thus increasing "talk range" (and effective coverage area).
Audio compression has long been used in two-way RF communications systems to increase "talk power" and effective range. See, for example, the following prior issued U.S. patents relating to audio compression amplifiers in RF systems:
U.S. Pat. No. 4,876,741 PA1 U.S. Pat. No. 4,718,116 PA1 U.S. Pat. No. 4,539,707 PA1 U.S. Pat. No. 4,381,488 PA1 U.S. Pat. No. 4,323,731 PA1 U.S. Pat. No. 4,216,427 PA1 U.S. Pat. No. 4,110,699 PA1 U.S. Pat. No. 3,449,684.
Ericsson-GE has in the past used analog compressors in its MASTR II Base Stations (see EGE maintenance manual LBI-30705) and in some remote control repeaters and transceivers.
In addition, a highly-specialized analog compression amplifier model 4006 Amplifier has long been included in EGE SimulCast repeater systems to maintain high average modulation level. See TELLABS technical manual 76-814006 for a description of such compression amplifier.
Generally, a compression amplifier must provide a variable gain (or attenuation) factor that is a function of input signal amplitude to provide a relatively constant output amplitude level that is substantially unaffected by input level variations. By varying the amount of gain (or attenuation) applied to the audio input when operating in a compression region, the compression amplifier can produce an output signal having a relatively uniform output level (e.g., for all input signals above a certain level).
But there are significant complexities involved in compressing audio signals. The selection of the gain factor applied by a compression amplifier typically depends on the average amplitude of the audio line signal. However, human speech is characterized by periods of activity interspersed with periods of silence. It is difficult or impossible to accurately determine an average amplitude at the beginning of an utterance, for example. The beginning of a spoken sentence is marked by silence followed by a nearly instantaneous increase in amplitude. During this brief period of time, the compressor must somehow determine an appropriate gain factor--even though it has very little information from which to set such gain factor.
Further, human speech includes many short duration bursts having high amplitudes between which are interspersed much lower amplitude speech components and pauses between phonemes. Processing a normal speech signal to generate an output signal of uniform amplitude despite variations in input level can result in loss of intelligibility (e.g., by causing background noise during pauses to be amplified to the level of the loudest speech components), and can introduce very unnatural sounding and distracting effects (e.g., "pumping", in which the background noise level "pumps" up and down in response to speech levels, and also "pop" noises due to discontinuous signals into a speaker).
To avoid such problems, most compressors designed for processing human speech provide a "fast attack, slow decay" response characteristic. Such amplifiers respond to the time average of input signal amplitude (rather than instantaneous input signal amplitude), and provide an asymmetrical gain response characteristic (i.e., they respond more rapidly to increase in input signal amplitude than they respond to decrease in signal amplitude). Attack time is the time it takes to settle to the proper output level when an increase in input signal amplitude occurs. Fast attack is important to prevent the first few phonemes from being noticeably louder than the following speech (causing unnatural sounding speech, and in RF systems, causing overmodulation). Slower release time is important to avoid bringing up the level of background noise during pauses between phonemes. The attack and release times should allow the compressor function to be as transparent as possible by not introducing annoying and abnormal level variations.
Prior hard-wired analog amplifier circuits provide a compressing variable gain transfer function by using a feedback network that adjusts amplifier gain depending on an average (over time) of the input signal level. A typical example of a prior art analog compression circuit is shown in the "Linear LSI Data and Applications Manual 1985" published by Signetics. This manual describes a "compandor" integrated circuit type NE570/571 that can be used as a basic compressor. Such a circuit can be used to provide "fast attack, slow release", and can also be configured to provide limiting action. A hard limiter is commonly associated with the attack time of the compressor. The limiting function prevents the compressor output signal from exceeding a predetermined level. For example, if the level of a signal suddenly increases, the compressor may apply too much gain to the signal, and a listener may hear a brief loud audio burst during this time. Such loud bursts are irritating to the listener. A hard limiter prevents these loud bursts from reaching the output (and in RF systems, prevents the transmitter from being overmodulated in violation of FCC regulations). Prior limiters have been implemented with comparators that ensure that the compressor output signal does not exceed a predetermined level.
Generally, a "fast attack, slow release" response characteristic is provided by using different RC networks. One of the networks (i.e., the one responsible for the attack gain) is effectively switched into circuit when a significant increase in input signal amplitude occurs, and provides a signal power integrator (i.e., a time averaging function) having a time constant associated with attack. The other RC network (i.e., the one responsible for release) is switched into circuit when a significant decrease in input signal amplitude occurs, and provides a signal power integrator having a time constant associated with release (the release time constant is usually much longer than the attack time constant). Such switching may be performed automatically by diode switches. Unfortunately, to alter the attack and release characteristics of such compression amplifiers, it is typically necessary to replace the RF network components.
Prior analog compressors have several shortcomings which can be generally categorized as inflexibility, slow response, circuit complexity, and non-ideal characteristics. Their operating parameters are inflexible and, as mentioned above, the attack and release parameters are fixed (changes in such parameters required component changes on the circuit board). For example, the EGE MASTR II prior art compressor applied a fixed finite change in the audio output that was a function of the input (previously, the change in the output was specified at 3 dB for a 30 dB change in the input). This compressor included a manual potentiometer that set the threshold for line input level at which the compressor began to compress, and used RC networks to provide attack and release signal integration over time. The compressor adjusted input signal gain only when the input signal exceeds the threshold (and otherwise provided linear amplification).
Prior art analog compressors are also generally slow to respond to changes in the audio input signal. The response times of RC networks and associated op amps limit the ranges of attack and release times the compressor may provide. Moreover, since many tradeoffs must be made in designing a hardware analog compressor, such compressors do not exhibit optimal signal processing characteristics. For example, analog compressor circuits do not produce a perfect flatness of input/output ratio over all frequency and other input signal variations. The dynamic performance of these compressors causes the input/output ratio to vary over the audio spectrum, introducing distortion. Moreover, the performance characteristics of analog compressors can be affected by environmental factors, aging and temperature, for example.
Analog compressors also employ relatively complex circuitry. These compressors have many components that increase the amount of circuit board "real estate" required, make circuit troubleshooting difficult, and increasing labor and parts costs associated with assembly and with maintaining manufacturing and replacement parts inventory.
It is generally known to use digital signal processing techniques to provide certain companding functions (e.g., for telephone applications). See, for example, U.S. Pat. No. 4,839,906 to Leveque et al. entitled "Processor Based Linked Compressor-Expander Telecommunications System" (1989); and U.S. Pat. No. 4,809,274 to Walker et al. entitled "Digital Audio Companding and Error Conditioning" (1989). Such systems convert an audio signal to a corresponding digital signal using an analog-to-digital converter; compress the signal in the digital domain (multiplying a digital value corresponds to gain adjustment in the analog domain); and convert the compressed digital signal to an analog signal using an digital-to-analog converter.
Much additional flexibility is provided by using digital signal processing to effect signal amplitude (dynamic range) compression. For example, compressor characteristics can be changed merely by effecting software changes (e.g., reloading or replacing an EEPROM) without necessitating hardware changes. In addition, compression can more easily be combined with other audio processing (e.g., filtering) without increasing parts count or circuit complexity.
The present invention relates to a RF base station system that uses audio compression to accommodate multiple audio line inputs. Two or more dispatchers (remote units) at different locations can be connected to a common base station over the same land line. The RF base station receives and processes audio signals from the multiple remote units simultaneously. The present invention ensures that each participant in a given RF communication (e.g., multiple land-based dispatchers and multiple RF transceiver users in the field) can hear all other participants without any one participant being louder than any other.
The RF base station provided by the present invention compresses a sum of multiple audio signals using a digital signal processor compressor so that all signals within the sum are equalized to a substantially uniform audio output level--despite differences in their input levels. Moreover, the RF base station provided by the present invention provides such compression in a manner that permits compression thresholds to be adjusted from a remote location. The RF base station provided by the present invention provides audio signal compression using digital signal processing techniques such that advanced features and flexibility are built into the system.
The present invention provides, in accordance with one of its important aspects, a digital, programmable compressor based on digital signal processing technology. In the preferred embodiment, line audio inputs from several sources are filtered, converted to digital signals, and processed by a digital signal processor (DSP). This DSP provides an audio compression function that selects and applies appropriate gains to the digitized audio signal. The compressed signal is converted to analog form, filtered and routed to a destination such as a RF transmitter modulator and/or a land line.
The present digital programmable compressor performs signal processing functions on the digitized audio signal in addition to amplitude compression so as to provide an enhanced output signal especially suited for RF transmission. For example, a noise blanking function is provided to minimize background noise. In the preferred embodiment, the DSP continually checks the input signal level, and outputs a zero (0) level if the input voltage is below a threshold value. Thus, the usual background hum and noise from audio lines are eliminated.
The compressor provided by the present invention also provides hard limiting to prevent the compressor output signal amplitude from exceeding a selected value. The automatic gain control function of the compressor is achieved to level multiple line inputs. Moreover, the present invention provides digital control (and remote setting) of the various thresholds and other control parameters set in the compressor. For example, the compressor includes software "potentiometers" that can be set by the digital signal processor. The compressor thresholds can be modified by resetting these "potentiometers."
The present invention provides additional flexibility in the operation of a compressor that allows the operating parameters and other performance characteristics to be controlled by a programmable processor. The present invention also provides optimal compression of input audio signals to generate an output signal which, for example, exhibits a substantially flat input/output signal ratio over a desired range. Moreover, another objective of the invention is to compress audio signals with an uncomplicated digital signal processing circuit that is relatively inexpensive to manufacture, operate and maintain.